lavc: AV-prefix all codec flags
[libav.git] / libavcodec / aacenc.c
1 /*
2 * AAC encoder
3 * Copyright (C) 2008 Konstantin Shishkov
4 *
5 * This file is part of Libav.
6 *
7 * Libav is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * Libav is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22 /**
23 * @file
24 * AAC encoder
25 */
26
27 /***********************************
28 * TODOs:
29 * add sane pulse detection
30 * add temporal noise shaping
31 ***********************************/
32
33 #include "libavutil/float_dsp.h"
34 #include "libavutil/opt.h"
35 #include "avcodec.h"
36 #include "put_bits.h"
37 #include "internal.h"
38 #include "mpeg4audio.h"
39 #include "kbdwin.h"
40 #include "sinewin.h"
41
42 #include "aac.h"
43 #include "aactab.h"
44 #include "aacenc.h"
45
46 #include "psymodel.h"
47
48 #define AAC_MAX_CHANNELS 6
49
50 #define ERROR_IF(cond, ...) \
51 if (cond) { \
52 av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
53 return AVERROR(EINVAL); \
54 }
55
56 float ff_aac_pow34sf_tab[428];
57
58 static const uint8_t swb_size_1024_96[] = {
59 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
60 12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
61 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
62 };
63
64 static const uint8_t swb_size_1024_64[] = {
65 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
66 12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
67 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
68 };
69
70 static const uint8_t swb_size_1024_48[] = {
71 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
72 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
73 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
74 96
75 };
76
77 static const uint8_t swb_size_1024_32[] = {
78 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
79 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
80 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
81 };
82
83 static const uint8_t swb_size_1024_24[] = {
84 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
85 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
86 32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
87 };
88
89 static const uint8_t swb_size_1024_16[] = {
90 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
91 12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
92 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
93 };
94
95 static const uint8_t swb_size_1024_8[] = {
96 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
97 16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
98 32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
99 };
100
101 static const uint8_t *swb_size_1024[] = {
102 swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
103 swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
104 swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
105 swb_size_1024_16, swb_size_1024_16, swb_size_1024_8
106 };
107
108 static const uint8_t swb_size_128_96[] = {
109 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
110 };
111
112 static const uint8_t swb_size_128_48[] = {
113 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
114 };
115
116 static const uint8_t swb_size_128_24[] = {
117 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
118 };
119
120 static const uint8_t swb_size_128_16[] = {
121 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
122 };
123
124 static const uint8_t swb_size_128_8[] = {
125 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
126 };
127
128 static const uint8_t *swb_size_128[] = {
129 /* the last entry on the following row is swb_size_128_64 but is a
130 duplicate of swb_size_128_96 */
131 swb_size_128_96, swb_size_128_96, swb_size_128_96,
132 swb_size_128_48, swb_size_128_48, swb_size_128_48,
133 swb_size_128_24, swb_size_128_24, swb_size_128_16,
134 swb_size_128_16, swb_size_128_16, swb_size_128_8
135 };
136
137 /** default channel configurations */
138 static const uint8_t aac_chan_configs[6][5] = {
139 {1, TYPE_SCE}, // 1 channel - single channel element
140 {1, TYPE_CPE}, // 2 channels - channel pair
141 {2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo
142 {3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center
143 {3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo
144 {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
145 };
146
147 /**
148 * Table to remap channels from Libav's default order to AAC order.
149 */
150 static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
151 { 0 },
152 { 0, 1 },
153 { 2, 0, 1 },
154 { 2, 0, 1, 3 },
155 { 2, 0, 1, 3, 4 },
156 { 2, 0, 1, 4, 5, 3 },
157 };
158
159 /**
160 * Make AAC audio config object.
161 * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
162 */
163 static void put_audio_specific_config(AVCodecContext *avctx)
164 {
165 PutBitContext pb;
166 AACEncContext *s = avctx->priv_data;
167
168 init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
169 put_bits(&pb, 5, 2); //object type - AAC-LC
170 put_bits(&pb, 4, s->samplerate_index); //sample rate index
171 put_bits(&pb, 4, s->channels);
172 //GASpecificConfig
173 put_bits(&pb, 1, 0); //frame length - 1024 samples
174 put_bits(&pb, 1, 0); //does not depend on core coder
175 put_bits(&pb, 1, 0); //is not extension
176
177 //Explicitly Mark SBR absent
178 put_bits(&pb, 11, 0x2b7); //sync extension
179 put_bits(&pb, 5, AOT_SBR);
180 put_bits(&pb, 1, 0);
181 flush_put_bits(&pb);
182 }
183
184 #define WINDOW_FUNC(type) \
185 static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
186 SingleChannelElement *sce, \
187 const float *audio)
188
189 WINDOW_FUNC(only_long)
190 {
191 const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
192 const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
193 float *out = sce->ret_buf;
194
195 fdsp->vector_fmul (out, audio, lwindow, 1024);
196 fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
197 }
198
199 WINDOW_FUNC(long_start)
200 {
201 const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
202 const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
203 float *out = sce->ret_buf;
204
205 fdsp->vector_fmul(out, audio, lwindow, 1024);
206 memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
207 fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
208 memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
209 }
210
211 WINDOW_FUNC(long_stop)
212 {
213 const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
214 const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
215 float *out = sce->ret_buf;
216
217 memset(out, 0, sizeof(out[0]) * 448);
218 fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
219 memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
220 fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
221 }
222
223 WINDOW_FUNC(eight_short)
224 {
225 const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
226 const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
227 const float *in = audio + 448;
228 float *out = sce->ret_buf;
229 int w;
230
231 for (w = 0; w < 8; w++) {
232 fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
233 out += 128;
234 in += 128;
235 fdsp->vector_fmul_reverse(out, in, swindow, 128);
236 out += 128;
237 }
238 }
239
240 static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
241 SingleChannelElement *sce,
242 const float *audio) = {
243 [ONLY_LONG_SEQUENCE] = apply_only_long_window,
244 [LONG_START_SEQUENCE] = apply_long_start_window,
245 [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
246 [LONG_STOP_SEQUENCE] = apply_long_stop_window
247 };
248
249 static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
250 float *audio)
251 {
252 int i;
253 float *output = sce->ret_buf;
254
255 apply_window[sce->ics.window_sequence[0]](&s->fdsp, sce, audio);
256
257 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
258 s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
259 else
260 for (i = 0; i < 1024; i += 128)
261 s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2);
262 memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
263 }
264
265 /**
266 * Encode ics_info element.
267 * @see Table 4.6 (syntax of ics_info)
268 */
269 static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
270 {
271 int w;
272
273 put_bits(&s->pb, 1, 0); // ics_reserved bit
274 put_bits(&s->pb, 2, info->window_sequence[0]);
275 put_bits(&s->pb, 1, info->use_kb_window[0]);
276 if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
277 put_bits(&s->pb, 6, info->max_sfb);
278 put_bits(&s->pb, 1, 0); // no prediction
279 } else {
280 put_bits(&s->pb, 4, info->max_sfb);
281 for (w = 1; w < 8; w++)
282 put_bits(&s->pb, 1, !info->group_len[w]);
283 }
284 }
285
286 /**
287 * Encode MS data.
288 * @see 4.6.8.1 "Joint Coding - M/S Stereo"
289 */
290 static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
291 {
292 int i, w;
293
294 put_bits(pb, 2, cpe->ms_mode);
295 if (cpe->ms_mode == 1)
296 for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
297 for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
298 put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
299 }
300
301 /**
302 * Produce integer coefficients from scalefactors provided by the model.
303 */
304 static void adjust_frame_information(ChannelElement *cpe, int chans)
305 {
306 int i, w, w2, g, ch;
307 int start, maxsfb, cmaxsfb;
308
309 for (ch = 0; ch < chans; ch++) {
310 IndividualChannelStream *ics = &cpe->ch[ch].ics;
311 start = 0;
312 maxsfb = 0;
313 cpe->ch[ch].pulse.num_pulse = 0;
314 for (w = 0; w < ics->num_windows*16; w += 16) {
315 for (g = 0; g < ics->num_swb; g++) {
316 //apply M/S
317 if (cpe->common_window && !ch && cpe->ms_mask[w + g]) {
318 for (i = 0; i < ics->swb_sizes[g]; i++) {
319 cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
320 cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
321 }
322 }
323 start += ics->swb_sizes[g];
324 }
325 for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--)
326 ;
327 maxsfb = FFMAX(maxsfb, cmaxsfb);
328 }
329 ics->max_sfb = maxsfb;
330
331 //adjust zero bands for window groups
332 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
333 for (g = 0; g < ics->max_sfb; g++) {
334 i = 1;
335 for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
336 if (!cpe->ch[ch].zeroes[w2*16 + g]) {
337 i = 0;
338 break;
339 }
340 }
341 cpe->ch[ch].zeroes[w*16 + g] = i;
342 }
343 }
344 }
345
346 if (chans > 1 && cpe->common_window) {
347 IndividualChannelStream *ics0 = &cpe->ch[0].ics;
348 IndividualChannelStream *ics1 = &cpe->ch[1].ics;
349 int msc = 0;
350 ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
351 ics1->max_sfb = ics0->max_sfb;
352 for (w = 0; w < ics0->num_windows*16; w += 16)
353 for (i = 0; i < ics0->max_sfb; i++)
354 if (cpe->ms_mask[w+i])
355 msc++;
356 if (msc == 0 || ics0->max_sfb == 0)
357 cpe->ms_mode = 0;
358 else
359 cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
360 }
361 }
362
363 /**
364 * Encode scalefactor band coding type.
365 */
366 static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
367 {
368 int w;
369
370 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
371 s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
372 }
373
374 /**
375 * Encode scalefactors.
376 */
377 static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
378 SingleChannelElement *sce)
379 {
380 int off = sce->sf_idx[0], diff;
381 int i, w;
382
383 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
384 for (i = 0; i < sce->ics.max_sfb; i++) {
385 if (!sce->zeroes[w*16 + i]) {
386 diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
387 if (diff < 0 || diff > 120)
388 av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n");
389 off = sce->sf_idx[w*16 + i];
390 put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
391 }
392 }
393 }
394 }
395
396 /**
397 * Encode pulse data.
398 */
399 static void encode_pulses(AACEncContext *s, Pulse *pulse)
400 {
401 int i;
402
403 put_bits(&s->pb, 1, !!pulse->num_pulse);
404 if (!pulse->num_pulse)
405 return;
406
407 put_bits(&s->pb, 2, pulse->num_pulse - 1);
408 put_bits(&s->pb, 6, pulse->start);
409 for (i = 0; i < pulse->num_pulse; i++) {
410 put_bits(&s->pb, 5, pulse->pos[i]);
411 put_bits(&s->pb, 4, pulse->amp[i]);
412 }
413 }
414
415 /**
416 * Encode spectral coefficients processed by psychoacoustic model.
417 */
418 static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
419 {
420 int start, i, w, w2;
421
422 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
423 start = 0;
424 for (i = 0; i < sce->ics.max_sfb; i++) {
425 if (sce->zeroes[w*16 + i]) {
426 start += sce->ics.swb_sizes[i];
427 continue;
428 }
429 for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)
430 s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
431 sce->ics.swb_sizes[i],
432 sce->sf_idx[w*16 + i],
433 sce->band_type[w*16 + i],
434 s->lambda);
435 start += sce->ics.swb_sizes[i];
436 }
437 }
438 }
439
440 /**
441 * Encode one channel of audio data.
442 */
443 static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
444 SingleChannelElement *sce,
445 int common_window)
446 {
447 put_bits(&s->pb, 8, sce->sf_idx[0]);
448 if (!common_window)
449 put_ics_info(s, &sce->ics);
450 encode_band_info(s, sce);
451 encode_scale_factors(avctx, s, sce);
452 encode_pulses(s, &sce->pulse);
453 put_bits(&s->pb, 1, 0); //tns
454 put_bits(&s->pb, 1, 0); //ssr
455 encode_spectral_coeffs(s, sce);
456 return 0;
457 }
458
459 /**
460 * Write some auxiliary information about the created AAC file.
461 */
462 static void put_bitstream_info(AACEncContext *s, const char *name)
463 {
464 int i, namelen, padbits;
465
466 namelen = strlen(name) + 2;
467 put_bits(&s->pb, 3, TYPE_FIL);
468 put_bits(&s->pb, 4, FFMIN(namelen, 15));
469 if (namelen >= 15)
470 put_bits(&s->pb, 8, namelen - 14);
471 put_bits(&s->pb, 4, 0); //extension type - filler
472 padbits = -put_bits_count(&s->pb) & 7;
473 avpriv_align_put_bits(&s->pb);
474 for (i = 0; i < namelen - 2; i++)
475 put_bits(&s->pb, 8, name[i]);
476 put_bits(&s->pb, 12 - padbits, 0);
477 }
478
479 /*
480 * Copy input samples.
481 * Channels are reordered from Libav's default order to AAC order.
482 */
483 static void copy_input_samples(AACEncContext *s, const AVFrame *frame)
484 {
485 int ch;
486 int end = 2048 + (frame ? frame->nb_samples : 0);
487 const uint8_t *channel_map = aac_chan_maps[s->channels - 1];
488
489 /* copy and remap input samples */
490 for (ch = 0; ch < s->channels; ch++) {
491 /* copy last 1024 samples of previous frame to the start of the current frame */
492 memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
493
494 /* copy new samples and zero any remaining samples */
495 if (frame) {
496 memcpy(&s->planar_samples[ch][2048],
497 frame->extended_data[channel_map[ch]],
498 frame->nb_samples * sizeof(s->planar_samples[0][0]));
499 }
500 memset(&s->planar_samples[ch][end], 0,
501 (3072 - end) * sizeof(s->planar_samples[0][0]));
502 }
503 }
504
505 static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
506 const AVFrame *frame, int *got_packet_ptr)
507 {
508 AACEncContext *s = avctx->priv_data;
509 float **samples = s->planar_samples, *samples2, *la, *overlap;
510 ChannelElement *cpe;
511 int i, ch, w, g, chans, tag, start_ch, ret;
512 int chan_el_counter[4];
513 FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
514
515 if (s->last_frame == 2)
516 return 0;
517
518 /* add current frame to queue */
519 if (frame) {
520 if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
521 return ret;
522 }
523
524 copy_input_samples(s, frame);
525 if (s->psypp)
526 ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
527
528 if (!avctx->frame_number)
529 return 0;
530
531 start_ch = 0;
532 for (i = 0; i < s->chan_map[0]; i++) {
533 FFPsyWindowInfo* wi = windows + start_ch;
534 tag = s->chan_map[i+1];
535 chans = tag == TYPE_CPE ? 2 : 1;
536 cpe = &s->cpe[i];
537 for (ch = 0; ch < chans; ch++) {
538 IndividualChannelStream *ics = &cpe->ch[ch].ics;
539 int cur_channel = start_ch + ch;
540 overlap = &samples[cur_channel][0];
541 samples2 = overlap + 1024;
542 la = samples2 + (448+64);
543 if (!frame)
544 la = NULL;
545 if (tag == TYPE_LFE) {
546 wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
547 wi[ch].window_shape = 0;
548 wi[ch].num_windows = 1;
549 wi[ch].grouping[0] = 1;
550
551 /* Only the lowest 12 coefficients are used in a LFE channel.
552 * The expression below results in only the bottom 8 coefficients
553 * being used for 11.025kHz to 16kHz sample rates.
554 */
555 ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
556 } else {
557 wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel,
558 ics->window_sequence[0]);
559 }
560 ics->window_sequence[1] = ics->window_sequence[0];
561 ics->window_sequence[0] = wi[ch].window_type[0];
562 ics->use_kb_window[1] = ics->use_kb_window[0];
563 ics->use_kb_window[0] = wi[ch].window_shape;
564 ics->num_windows = wi[ch].num_windows;
565 ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
566 ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
567 for (w = 0; w < ics->num_windows; w++)
568 ics->group_len[w] = wi[ch].grouping[w];
569
570 apply_window_and_mdct(s, &cpe->ch[ch], overlap);
571 }
572 start_ch += chans;
573 }
574 if ((ret = ff_alloc_packet(avpkt, 768 * s->channels))) {
575 av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
576 return ret;
577 }
578
579 do {
580 int frame_bits;
581
582 init_put_bits(&s->pb, avpkt->data, avpkt->size);
583
584 if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT))
585 put_bitstream_info(s, LIBAVCODEC_IDENT);
586 start_ch = 0;
587 memset(chan_el_counter, 0, sizeof(chan_el_counter));
588 for (i = 0; i < s->chan_map[0]; i++) {
589 FFPsyWindowInfo* wi = windows + start_ch;
590 const float *coeffs[2];
591 tag = s->chan_map[i+1];
592 chans = tag == TYPE_CPE ? 2 : 1;
593 cpe = &s->cpe[i];
594 put_bits(&s->pb, 3, tag);
595 put_bits(&s->pb, 4, chan_el_counter[tag]++);
596 for (ch = 0; ch < chans; ch++)
597 coeffs[ch] = cpe->ch[ch].coeffs;
598 s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
599 for (ch = 0; ch < chans; ch++) {
600 s->cur_channel = start_ch + ch;
601 s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
602 }
603 cpe->common_window = 0;
604 if (chans > 1
605 && wi[0].window_type[0] == wi[1].window_type[0]
606 && wi[0].window_shape == wi[1].window_shape) {
607
608 cpe->common_window = 1;
609 for (w = 0; w < wi[0].num_windows; w++) {
610 if (wi[0].grouping[w] != wi[1].grouping[w]) {
611 cpe->common_window = 0;
612 break;
613 }
614 }
615 }
616 s->cur_channel = start_ch;
617 if (s->options.stereo_mode && cpe->common_window) {
618 if (s->options.stereo_mode > 0) {
619 IndividualChannelStream *ics = &cpe->ch[0].ics;
620 for (w = 0; w < ics->num_windows; w += ics->group_len[w])
621 for (g = 0; g < ics->num_swb; g++)
622 cpe->ms_mask[w*16+g] = 1;
623 } else if (s->coder->search_for_ms) {
624 s->coder->search_for_ms(s, cpe, s->lambda);
625 }
626 }
627 adjust_frame_information(cpe, chans);
628 if (chans == 2) {
629 put_bits(&s->pb, 1, cpe->common_window);
630 if (cpe->common_window) {
631 put_ics_info(s, &cpe->ch[0].ics);
632 encode_ms_info(&s->pb, cpe);
633 }
634 }
635 for (ch = 0; ch < chans; ch++) {
636 s->cur_channel = start_ch + ch;
637 encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);
638 }
639 start_ch += chans;
640 }
641
642 frame_bits = put_bits_count(&s->pb);
643 if (frame_bits <= 6144 * s->channels - 3) {
644 s->psy.bitres.bits = frame_bits / s->channels;
645 break;
646 }
647
648 s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;
649
650 } while (1);
651
652 put_bits(&s->pb, 3, TYPE_END);
653 flush_put_bits(&s->pb);
654 avctx->frame_bits = put_bits_count(&s->pb);
655
656 // rate control stuff
657 if (!(avctx->flags & AV_CODEC_FLAG_QSCALE)) {
658 float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
659 s->lambda *= ratio;
660 s->lambda = FFMIN(s->lambda, 65536.f);
661 }
662
663 if (!frame)
664 s->last_frame++;
665
666 ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
667 &avpkt->duration);
668
669 avpkt->size = put_bits_count(&s->pb) >> 3;
670 *got_packet_ptr = 1;
671 return 0;
672 }
673
674 static av_cold int aac_encode_end(AVCodecContext *avctx)
675 {
676 AACEncContext *s = avctx->priv_data;
677
678 ff_mdct_end(&s->mdct1024);
679 ff_mdct_end(&s->mdct128);
680 ff_psy_end(&s->psy);
681 if (s->psypp)
682 ff_psy_preprocess_end(s->psypp);
683 av_freep(&s->buffer.samples);
684 av_freep(&s->cpe);
685 ff_af_queue_close(&s->afq);
686 return 0;
687 }
688
689 static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
690 {
691 int ret = 0;
692
693 avpriv_float_dsp_init(&s->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
694
695 // window init
696 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
697 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
698 ff_init_ff_sine_windows(10);
699 ff_init_ff_sine_windows(7);
700
701 if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0))
702 return ret;
703 if (ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0))
704 return ret;
705
706 return 0;
707 }
708
709 static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
710 {
711 int ch;
712 FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 3 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail);
713 FF_ALLOCZ_OR_GOTO(avctx, s->cpe, sizeof(ChannelElement) * s->chan_map[0], alloc_fail);
714 FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
715
716 for(ch = 0; ch < s->channels; ch++)
717 s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
718
719 return 0;
720 alloc_fail:
721 return AVERROR(ENOMEM);
722 }
723
724 static av_cold int aac_encode_init(AVCodecContext *avctx)
725 {
726 AACEncContext *s = avctx->priv_data;
727 int i, ret = 0;
728 const uint8_t *sizes[2];
729 uint8_t grouping[AAC_MAX_CHANNELS];
730 int lengths[2];
731
732 avctx->frame_size = 1024;
733
734 for (i = 0; i < 16; i++)
735 if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
736 break;
737
738 s->channels = avctx->channels;
739
740 ERROR_IF(i == 16,
741 "Unsupported sample rate %d\n", avctx->sample_rate);
742 ERROR_IF(s->channels > AAC_MAX_CHANNELS,
743 "Unsupported number of channels: %d\n", s->channels);
744 ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW,
745 "Unsupported profile %d\n", avctx->profile);
746 ERROR_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
747 "Too many bits per frame requested\n");
748
749 s->samplerate_index = i;
750
751 s->chan_map = aac_chan_configs[s->channels-1];
752
753 if ((ret = dsp_init(avctx, s)) < 0)
754 goto fail;
755
756 if ((ret = alloc_buffers(avctx, s)) < 0)
757 goto fail;
758
759 avctx->extradata_size = 5;
760 put_audio_specific_config(avctx);
761
762 sizes[0] = swb_size_1024[i];
763 sizes[1] = swb_size_128[i];
764 lengths[0] = ff_aac_num_swb_1024[i];
765 lengths[1] = ff_aac_num_swb_128[i];
766 for (i = 0; i < s->chan_map[0]; i++)
767 grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
768 if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
769 s->chan_map[0], grouping)) < 0)
770 goto fail;
771 s->psypp = ff_psy_preprocess_init(avctx);
772 s->coder = &ff_aac_coders[2];
773
774 s->lambda = avctx->global_quality ? avctx->global_quality : 120;
775
776 ff_aac_tableinit();
777
778 for (i = 0; i < 428; i++)
779 ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
780
781 avctx->initial_padding = 1024;
782 ff_af_queue_init(avctx, &s->afq);
783
784 return 0;
785 fail:
786 aac_encode_end(avctx);
787 return ret;
788 }
789
790 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
791 static const AVOption aacenc_options[] = {
792 {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.i64 = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
793 {"auto", "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
794 {"ms_off", "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
795 {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
796 {NULL}
797 };
798
799 static const AVClass aacenc_class = {
800 "AAC encoder",
801 av_default_item_name,
802 aacenc_options,
803 LIBAVUTIL_VERSION_INT,
804 };
805
806 AVCodec ff_aac_encoder = {
807 .name = "aac",
808 .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
809 .type = AVMEDIA_TYPE_AUDIO,
810 .id = AV_CODEC_ID_AAC,
811 .priv_data_size = sizeof(AACEncContext),
812 .init = aac_encode_init,
813 .encode2 = aac_encode_frame,
814 .close = aac_encode_end,
815 .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
816 CODEC_CAP_EXPERIMENTAL,
817 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
818 AV_SAMPLE_FMT_NONE },
819 .priv_class = &aacenc_class,
820 };